I'm in the external hardware camp as well. I like a small analog mixer with built in converters. For a long time I had a mackie 1202 onyx with a firewire card (I made the mistake of buying a machine which would not run win7 and win10 obsoleted firewire) now I have a presonous ar16usb. My rig can sit around for weeks at a time but both of those setups are so simple and reliable I can fire it up and be recording tracks if I choose within a couple of minutes without having to remember or navigate a bunch of things.
A footnote is I 'think' the onyx sounds slightly better. That said, I just use 44.1khz sample rate and I suspect the decimation filter in the board favors higher sample rates. One day I'll try higher rates and see if I can perceive a difference. On the other hand, I may be responding to differences between the relative shapes of the filters and not a digital artifact (which would result from frequencies greater than 22050 hz leaking into the converter). I have not attempted to do a side by side - I just get the sense of a slight and subtle high frequency harshness.
The presonus has a few tricks there is a superchannel which I thought I would never use that has several inputs and lets you pair bluetooth but I have already put it into use several times.
John
BTW, I referred to the filter before the analog to digital converter with the wrong name. It is the anti-aliasing filter.
The anti-aliasing filter brings to mind a couple of questions.
Are these generally integrated into a ADC chip?
Do the ADCs usually sample at a high rate and down-convert for slower sample-rates?
I realize there are a lot of methods that can be employed, but there is possibly just a handful of ADCs used in a majority of pro-audio gear.
What I do know about these filters (although its been 20 years or so since I read about them) is they are analog lowpass filters which ideally have a flat frequency response in the audible range and a very sharp cutoff curve above that. Of course they are not quite ideal in the real world, where the sharper the cutoff, the more artifacts introduced like ripples in response and phase shifting of some frequencies.
Bob,
...Just watched the "SAWStudio Band Session Part 1" video and this does indeed show how to configure the instruments.Watch one of my old band recording videos on the website... it shows how to assign the input sources and arm the tracks and overdub.
Just two questions then...
Q1: Is the [Input Source] for a given track simply left at [MultiTrack] then?
Q2: What is the purpose then of the [Input Source] being set to a [Device]?
Thanks in advance...
PS: Great vids by-the -way!!
You can set input sources on tracks to a device input directly if you want to use the channel strip processing as part of the recording...
In most cases, you are better off recording the signal with no input processing and process the signal during mixdown later for more flexibility... if you capture the signal with eq or compression... you are stuck with that processing... if you capture flat... then you can alter the signal in many various ways after the recording.
Bob L
Bob,
... I understand..makes sense!You can set input sources on tracks to a device input directly if you want to use the channel strip processing as part of the recording...
... I understand..makes sense as well!... if you capture flat... then you can alter the signal in many various ways after the recording.
Thank, Bob.
Ah! I'm sure I knew that ten years ago.
Bob -- where's the monitoring "tap" point for tape-style monitoring? Just wondering if/when any FX plugins' latency might impact overdubbing when using a direct device input on a recorded track. Or is that magically compensated for even in said scenario?
Dave "it aint the heat, it's the humidity" Labrecque
Becket, Massachusetts
The record tap for device input sources is direct from the device input... plugins will be involved in the playback portion for the overdub... really not sure if plugin latency will affect things... I believe it should not because the playback itself should be compensated.
Bob L
You could test it by 'overdubbing' on the playback of a click track processed through latency plugins and re-recorded into your mic (from one headphone inverted out). In reality, I've never had a problem overdubbing in sync... but with punch-ins just make sure buffers are not set too high.
Also, as Bob mentioned, there is so much more mix flexibility in playback when recording dry rather than recording a preprocessed recorded track (whether out of the box or in the box).
Last edited by Carl G.; 06-02-2020 at 02:00 PM.
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